Each year the percentage of voice communications transmitted over packet-switched networks, such as the Internet, is increasing while the percentage of voice communications transmitted over traditional circuit-switched networks, such as the Public Switch Telecommunications Network or PSTN, is decreasing. Substantial cost savings, particularly for long distance calls, can be realized using Voice Over IP or VoIP when compared to PSTN calls. A problem with this transition to packet-switched networks is maintaining a high Quality of Service or QoS notwithstanding higher levels of traffic across the network. To provide the continuous data stream required for real-time phone calls and video-conferencing, certain QoS metrics, particularly packet latency, must be kept low with a low variance throughout the many switching and routing hops the packets may encounter. As will be appreciated, “packet latency” or “packet delay” refers to the length of time required for a packet of data to travel from one designated point to another. While voice and video quality is certainly in the user's interest, it also provides surprising benefits for service providers. Studies done of cellular networks show that, as voice quality increases, the time users spend on the network also increases, which means billable hours—and thus revenues—increase too.
One approach used to provide enhanced QoS metrics is to encode the voice stream. Various compression/decompression algorithms (or codecs) have been developed. The codecs are able to take an audio stream occupying a first bandwidth and compress it to occupy a much smaller second bandwidth. As will be appreciated, “bandwidth” refers to a measure of the data transmission capacity of a network or a component thereof and is usually expressed in how many bits of data can be moved from one point to another in a unit of time under ideal conditions. Examples of codecs are as follows:
ITU Codec StandardBit RateG.711 (Pulse Code Modulation)3.4 kHz at 56 or 64 KbpsG.722 (audio codec)7 kHz at anywhere between 48,56 or 64 KbpsG.722.1 (wideband)7 kHz at 24 or 32 KbpsG.7233.4 kHz at 24 KbpsG.723.13.4 kHz at 5.3 or 6.4 KbpsG.726, G.727 (Adaptive Differential4 kHz at 16, 24, 32, or 40 KbpsPulse Code Modulation (ADPCM)G.728 (audio codec)3 kHz at 4 or 16 KbpsG.729 (speech codec)3.4 kHz at 8 KbpsAs will be appreciated, a greater the degree of compression provides generally a lower degree of voice quality.
Another QoS approach is employed by the H.323 and ReSerVation (or RSVP) Protocols. RSVP reserves network resources for a sender. Senders first specify their outgoing traffic in terms of the preferred upper and lower bounds of bandwidth, delay, and jitter. As will be appreciated, “jitter” refers to a distortion of the interpacket arrival times (the interval between packet arrivals). The network maintains this flow specification, which describes both the source's traffic stream and the application's service requirements. Although RSVP can provide acceptable QoS metrics on a successfully negotiated (reserved) path, RSVP suffers from an inability to scale over large networks. If along some network path a hop will not or does not provide the QoS requested by RSVP, the entire path is deemed inadequate. Moreover, a successfully negotiated RSVP path is problematical in that RSVP requires unshared dedicated bandwidth and imposes a substantial processing overhead on all networking elements.
Call admission or bandwidth control is another QoS solution employed to provide improved transmission characteristics end-to-end, such as available bandwidth, maximum end-to-end delay, maximum end-to-end delay variation (jitter), and maximum packet/cell loss. Call admission control, which is typically applied at a gatekeeper or gateway, regulates audio quality by limiting the number of VoIP calls that can be active on a particular link at the same time. Existing QoS measurement based call admission control mechanisms are utilization-threshold based or response-time-response probes. In utilization-threshold based call admission control, the call admission controller monitors one or more selected links and controls call placement depending on the available bandwidth on the link(s) and the required bandwidth for the call. For example, if each call requires a bandwidth of 24 kilobits/second (for a G.723 codec) and the available bandwidth for voice calls is 100 kilobits/second, call admission control will allow up to four calls to be placed and block any additional calls from being made while the bandwidth is in use. In response-time-response probe call admission control, when a call placement request is received the call admission controller determines, using measurements from various probes distributed over the network and/or network pings, the current QoS metrics in the network and, based on the metrics and the codec to be used for the call, does not permit placement of the call when desired or specified QoS metrics will not be satisfied for the call. As will be appreciated, in the absence of call admission control the placement of a call when the available bandwidth is already being used to capacity can not only provide a low audio quality for the newly placed call but also provide a lower audio quality for other calls using the congested link.
Call admission control can become problematic because the available bandwidth/allowable call quota are static variables and not dynamically optimized in response to network state. The use of static variables works very well if a link has been provisioned for VoIP and only one type of codec is used for all calls. When a combination of a high bit rate and low bit rate codecs are used, it can become difficult to exactly specify how many sessions a link can support. This can occur when the destination requests a codec different from the codec requested by the source. The use of signals from probes and pings to measure the state of the network can also be problematic. Such signals can be serviced last or completely dropped during times of congestion. This behavior can provide a false negative on network performance. This approach also may not provide detailed feedback that would indicate that the network state can descend into ill performance if more traffic is offered, thereby eliminating the possibility of pre-emptive corrective measures being undertaken. By the time that the signal measurements are processed, the VoIP QoS can already be affected. An alternative is to set the thresholds, against which the measurements are compared, very low. This approach, however, may have the adverse effect of calls being unnecessarily blocked during times of temporary congestion.
A brute force solution to the problem of insufficient bandwidth is simply to use more bandwidth. Using telephony traffic formulas, one can ascertain the maximum number of simultaneous calls during the busiest period. The bandwidth requirements for this number of calls would be calculated and then a Wide Area Network or WAN link with more available bandwidth than maximally needed obtained. This approach does solve one problem, namely that of limited bandwidth, while leading to another, namely that of undue expense. Bandwidth is expensive, especially WAN bandwidth.